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https://github.com/mollyim/webrtc.git
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0f030fd263
ChannelReceive for audio has both a thread checker and pointer. Both aren't needed, so this removes the checker. Moving forward we should be able to guard more variables with checks and remove the need for locks. Removing module_process_thread_checker_ from AudioReceiveStream. The checker was misleading and actually checked the worker thread. Updating downstream code in ChannelReceive accordingly. Bug: webrtc:11993 Change-Id: I93becd4989e5838412a4f079ba63cf67252daa84 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212613 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33616}
125 lines
4.5 KiB
C++
125 lines
4.5 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef AUDIO_AUDIO_RECEIVE_STREAM_H_
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#define AUDIO_AUDIO_RECEIVE_STREAM_H_
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#include <memory>
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#include <vector>
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#include "api/audio/audio_mixer.h"
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#include "api/neteq/neteq_factory.h"
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#include "api/rtp_headers.h"
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#include "api/sequence_checker.h"
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#include "audio/audio_state.h"
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#include "call/audio_receive_stream.h"
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#include "call/syncable.h"
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#include "modules/rtp_rtcp/source/source_tracker.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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class PacketRouter;
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class ProcessThread;
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class RtcEventLog;
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class RtpPacketReceived;
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class RtpStreamReceiverControllerInterface;
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class RtpStreamReceiverInterface;
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namespace voe {
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class ChannelReceiveInterface;
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} // namespace voe
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namespace internal {
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class AudioSendStream;
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class AudioReceiveStream final : public webrtc::AudioReceiveStream,
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public AudioMixer::Source,
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public Syncable {
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public:
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AudioReceiveStream(Clock* clock,
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RtpStreamReceiverControllerInterface* receiver_controller,
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PacketRouter* packet_router,
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ProcessThread* module_process_thread,
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NetEqFactory* neteq_factory,
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const webrtc::AudioReceiveStream::Config& config,
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const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
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webrtc::RtcEventLog* event_log);
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// For unit tests, which need to supply a mock channel receive.
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AudioReceiveStream(
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Clock* clock,
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RtpStreamReceiverControllerInterface* receiver_controller,
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PacketRouter* packet_router,
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const webrtc::AudioReceiveStream::Config& config,
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const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
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webrtc::RtcEventLog* event_log,
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std::unique_ptr<voe::ChannelReceiveInterface> channel_receive);
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AudioReceiveStream() = delete;
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AudioReceiveStream(const AudioReceiveStream&) = delete;
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AudioReceiveStream& operator=(const AudioReceiveStream&) = delete;
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~AudioReceiveStream() override;
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// webrtc::AudioReceiveStream implementation.
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void Reconfigure(const webrtc::AudioReceiveStream::Config& config) override;
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void Start() override;
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void Stop() override;
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bool IsRunning() const override;
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webrtc::AudioReceiveStream::Stats GetStats(
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bool get_and_clear_legacy_stats) const override;
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void SetSink(AudioSinkInterface* sink) override;
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void SetGain(float gain) override;
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bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
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int GetBaseMinimumPlayoutDelayMs() const override;
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std::vector<webrtc::RtpSource> GetSources() const override;
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// AudioMixer::Source
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AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
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AudioFrame* audio_frame) override;
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int Ssrc() const override;
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int PreferredSampleRate() const override;
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// Syncable
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uint32_t id() const override;
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absl::optional<Syncable::Info> GetInfo() const override;
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bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
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int64_t* time_ms) const override;
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void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
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int64_t time_ms) override;
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bool SetMinimumPlayoutDelay(int delay_ms) override;
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void AssociateSendStream(AudioSendStream* send_stream);
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void DeliverRtcp(const uint8_t* packet, size_t length);
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const webrtc::AudioReceiveStream::Config& config() const;
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const AudioSendStream* GetAssociatedSendStreamForTesting() const;
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private:
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static void ConfigureStream(AudioReceiveStream* stream,
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const Config& new_config,
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bool first_time);
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AudioState* audio_state() const;
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SequenceChecker worker_thread_checker_;
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webrtc::AudioReceiveStream::Config config_;
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rtc::scoped_refptr<webrtc::AudioState> audio_state_;
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SourceTracker source_tracker_;
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const std::unique_ptr<voe::ChannelReceiveInterface> channel_receive_;
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AudioSendStream* associated_send_stream_ = nullptr;
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bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false;
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std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_;
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};
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} // namespace internal
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} // namespace webrtc
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#endif // AUDIO_AUDIO_RECEIVE_STREAM_H_
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